FAQ: Basic VoIP Terms

If you aren't an IT wiz certain terms and concepts may seem a bit out of you comfort zone. But don't worry, we want to help you understand some simple VoIP terms. Let's explore a few basic VoIP terms to help you expand your knowledge.

What is Voice over IP (VoIP)?

Voice over IP, or VoIP for short, is a method and group of technologies for delivering voice and video communications over IP (Internet Protocol) networks, such as the internet or a company's local network.

What is SIP?

Session Initiation Protocol, or SIP, is a communications protocol used for signaling and controlling voice and video communications. It is the most widely-supported VoIP protocol. It does not 'carry' the packets of data that contain the digitized voice or video on a call but it handles signaling (ringing, hangup, etc.) and negotiates the voice or video codec that will be used by each endpoint.

What is a trunk?

A trunk, or more usually referred to as a 'SIP trunk', is a digital pathway used for making VoIP calls to or from the PSTN or another PBX.

Any call that is not made 'on net', or purely via IP networking, will need to interface with the PSTN and will need to use a trunk. All ITSP's provide a certain number of trunks to their customers' PBX systems, which determine the maximum number of concurrent calls to the PSTN that can be supported.

In older PBX technologies, a company might have a PRI with 23 'channels' for carring voice traffic, which would mean a maximum of 23 concurrent calls to the PSTN. Since these PSTN pathways are now based on IP networks, (i.e., SIP trunking), this means that call capacity is much more flexible and can be increased as needed on the fly, provided that other requisite services (bandwidth, PBX hardware specifications, etc.) are able to handle the increase. For companies that need multiple PBX systems, SIP trunks can also provide inter-PBX calling without traversing the PSTN and thus not incurring extra costs.

What is latency?

Latency refers to a delay in the delivery of data packets between two networks. In the VoIP world, latency is a core metric to consider when deploying a VoIP PBX. All audio and video that is IP based is sent in data packets across an IP network, like the internet, usually using RTP (Real-time Transport Protocol).

RTP 'streams' the packets to the endpoint without a mechanism for re-requesting a packet or packets should one be lost in transit.

If there is latency between the two networks, these packets can arrive out of order or be dropped altogether, resulting in poor call quality or even dropped calls.

Latency is usually measured in milliseconds and Virtualtone can provide a speedtest that will measure network latency for existing customers and for prequalification of potential customers.

One other note: Latency is indpendent of bandwidth. Bandwidth refers the capacity of a circuit to carry parallel data between networks. Bandwidth is measured in bits per second (bps), up to Gigabits per second (Gbps), and having a circuit with a large bandwidth does not guarantee a good customer experience with VoIP. Please contact Virtualtone about our speedtest or to get more information.

Want some more information about VirtualTone's VoIP communication system? Contact our Sales Team today to find out how we can take yoru business to the next level!

VirtualTone caters to all businesses from enterprise level call centers to small and mid-sized businesses. Our mission is to provide you with the highest quality communication solutions without pain or discomfort. VirtualTone serves clients in numerous industries, with experience in Insurance, Legal, Government, Construction, Oil & Gas, Hospitality and Manufacturing.